Merge branch 'master' into 0.11

common

@@ -1 +1 @@
-Subproject commit 46dfcea233cf6df83e3771d8a8066e87d614f893
+Subproject commit 69b981f10caa234ad0ff639179d0fda8505bd94b

configure.ac

@@ -369,7 +369,7 @@ else
 
      AS_MKDIR_P(["$ac_top_build_prefix"gst-libs/ext/libav])
      cd "$ac_top_build_prefix"gst-libs/ext/libav &&
-         $confcmd $embffmpeg_configure_args ||
+         eval "$confcmd $embffmpeg_configure_args" ||
          AC_MSG_ERROR([Failed to configure embedded Libav tree])
      cd "$origdir"
     ],

ext/ffmpeg/Makefile.am

@@ -22,7 +22,7 @@ libgstffmpeg_la_SOURCES = gstffmpeg.c	\
 # 			  gstffmpegscale.c
 
 libgstffmpeg_la_CFLAGS = $(FFMPEG_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
-libgstffmpeg_la_LIBADD = $(FFMPEG_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(LIBM) $(WIN32_LIBS) -lz $(BZ2_LIBS)
+libgstffmpeg_la_LIBADD = $(FFMPEG_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) -lgstpbutils-$(GST_MAJORMINOR) $(LIBM) $(WIN32_LIBS) -lz $(BZ2_LIBS)
 libgstffmpeg_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(DARWIN_LDFLAGS)
 libgstffmpeg_la_LIBTOOLFLAGS = --tag=disable-static
 

ext/ffmpeg/gstffmpegcodecmap.c

@@ -33,6 +33,8 @@
 #include "gstffmpeg.h"
 #include "gstffmpegcodecmap.h"
 
+#include <gst/pbutils/codec-utils.h>
+
 /*
  * Read a palette from a caps.
  */
@@ -439,6 +441,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
       case CODEC_ID_AC3:
       case CODEC_ID_EAC3:
       case CODEC_ID_AAC:
+      case CODEC_ID_AAC_LATM:
       case CODEC_ID_DTS:
         maxchannels = 6;
         break;
@@ -956,8 +959,52 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
       break;
 
     case CODEC_ID_AAC:
+    {
+      caps = gst_ff_aud_caps_new (context, codec_id, "audio/mpeg", NULL);
+
+      if (!encode) {
+        GValue arr = { 0, };
+        GValue item = { 0, };
+
+        g_value_init (&arr, GST_TYPE_LIST);
+        g_value_init (&item, G_TYPE_INT);
+        g_value_set_int (&item, 2);
+        gst_value_list_append_value (&arr, &item);
+        g_value_set_int (&item, 4);
+        gst_value_list_append_value (&arr, &item);
+        g_value_unset (&item);
+
+        gst_caps_set_value (caps, "mpegversion", &arr);
+        g_value_unset (&arr);
+
+        g_value_init (&arr, GST_TYPE_LIST);
+        g_value_init (&item, G_TYPE_STRING);
+        g_value_set_string (&item, "raw");
+        gst_value_list_append_value (&arr, &item);
+        g_value_set_string (&item, "adts");
+        gst_value_list_append_value (&arr, &item);
+        g_value_set_string (&item, "adif");
+        gst_value_list_append_value (&arr, &item);
+        g_value_unset (&item);
+
+        gst_caps_set_value (caps, "stream-format", &arr);
+        g_value_unset (&arr);
+      } else {
+        gst_caps_set_simple (caps, "mpegversion", G_TYPE_INT, 4,
+            "stream-format", G_TYPE_STRING, "raw",
+            "base-profile", G_TYPE_STRING, "lc", NULL);
+
+        if (context && context->extradata_size > 0)
+          gst_codec_utils_aac_caps_set_level_and_profile (caps,
+              context->extradata, context->extradata_size);
+      }
+
+      break;
+    }
+    case CODEC_ID_AAC_LATM:    /* LATM/LOAS AAC syntax */
       caps = gst_ff_aud_caps_new (context, codec_id, "audio/mpeg",
-          "mpegversion", G_TYPE_INT, 4, NULL);
+          "mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "loas",
+          NULL);
       break;
 
     case CODEC_ID_ASV1:

ext/ffmpeg/gstffmpegdec.c

@@ -822,8 +822,9 @@ gst_ffmpegdec_setcaps (GstPad * pad, GstCaps * caps)
     ffmpegdec->context->flags |= CODEC_FLAG_EMU_EDGE;
   }
 
-  /* for AAC we only use av_parse if not on raw caps */
+  /* for AAC we only use av_parse if not on stream-format==raw or ==loas */
-  if (oclass->in_plugin->id == CODEC_ID_AAC) {
+  if (oclass->in_plugin->id == CODEC_ID_AAC
+      || oclass->in_plugin->id == CODEC_ID_AAC_LATM) {
     const gchar *format = gst_structure_get_string (structure, "stream-format");
 
     if (format == NULL || strcmp (format, "raw") == 0) {
@@ -1423,14 +1424,7 @@ gst_ffmpegdec_do_qos (GstFFMpegDec * ffmpegdec, GstClockTime timestamp,
   if (proportion < 0.4 && diff < 0) {
     goto normal_mode;
   } else {
-    /* if we're more than two seconds late, switch to the next keyframe */
+    if (diff >= 0) {
-    /* FIXME, let the demuxer decide what's the best since we might be dropping
-     * a lot of frames when the keyframe is far away or we even might not get a new
-     * keyframe at all.. */
-    if (diff > ((GstClockTimeDiff) GST_SECOND * 2)
-        && !ffmpegdec->waiting_for_key) {
-      goto skip_to_keyframe;
-    } else if (diff >= 0) {
       /* we're too slow, try to speed up */
       if (ffmpegdec->waiting_for_key) {
         /* we were waiting for a keyframe, that's ok */
@@ -1457,16 +1451,6 @@ normal_mode:
     }
     return TRUE;
   }
-skip_to_keyframe:
-  {
-    ffmpegdec->context->skip_frame = AVDISCARD_NONKEY;
-    ffmpegdec->waiting_for_key = TRUE;
-    *mode_switch = TRUE;
-    GST_DEBUG_OBJECT (ffmpegdec,
-        "QOS: keyframe, %" G_GINT64_FORMAT " > GST_SECOND/2", diff);
-    /* we can skip the current frame */
-    return FALSE;
-  }
 skip_frame:
   {
     if (ffmpegdec->context->skip_frame != AVDISCARD_NONREF) {
@@ -1855,7 +1839,7 @@ gst_ffmpegdec_video_frame (GstFFMpegDec * ffmpegdec,
 
   /* we assume DTS as input timestamps unless we see reordered input
    * timestamps */
-  if (!ffmpegdec->reordered_in) {
+  if (!ffmpegdec->reordered_in && ffmpegdec->reordered_out) {
     /* PTS and DTS are the same for keyframes */
     if (!iskeyframe && ffmpegdec->next_out != -1) {
       /* interpolate all timestamps except for keyframes, FIXME, this is
@@ -2193,7 +2177,8 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
     GST_BUFFER_OFFSET (*outbuf) = out_offset;
 
     /* the next timestamp we'll use when interpolating */
-    ffmpegdec->next_out = out_timestamp + out_duration;
+    if (GST_CLOCK_TIME_IS_VALID (out_timestamp))
+      ffmpegdec->next_out = out_timestamp + out_duration;
 
     /* now see if we need to clip the buffer against the segment boundaries. */
     if (G_UNLIKELY (!clip_audio_buffer (ffmpegdec, *outbuf, out_timestamp,
@@ -2208,7 +2193,8 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
 
   /* If we don't error out after the first failed read with the AAC decoder,
    * we must *not* carry on pushing data, else we'll cause segfaults... */
-  if (len == -1 && in_plugin->id == CODEC_ID_AAC) {
+  if (len == -1 && (in_plugin->id == CODEC_ID_AAC
+          || in_plugin->id == CODEC_ID_AAC_LATM)) {
     GST_ELEMENT_ERROR (ffmpegdec, STREAM, DECODE, (NULL),
         ("Decoding of AAC stream by FFMPEG failed."));
     *ret = GST_FLOW_ERROR;
@@ -2666,7 +2652,7 @@ gst_ffmpegdec_chain (GstPad * pad, GstBuffer * inbuf)
           ffmpegdec->pctx->pts);
 
       /* store pts for decoding */
-      if (ffmpegdec->pctx->pts != -1)
+      if (ffmpegdec->pctx->pts != AV_NOPTS_VALUE && ffmpegdec->pctx->pts != -1)
         dec_info = gst_ts_info_get (ffmpegdec, ffmpegdec->pctx->pts);
       else {
         /* ffmpeg sometimes loses track after a flush, help it by feeding a

gst-libs/ext/libav

@@ -1 +1 @@
-Subproject commit 151cd4f98225c0c5cab8fb29e1dbc0719f7f67e3
+Subproject commit 371266daa3df35c424203fff0ce2e6de0e33a29d